I have been trying to play with my asterisk box and can’t get calls in or out. I am behind a NAT and have port forwarded to my box. My box is in a DMZ, I have turned off my firewall and have followed the Oreilly book to a tee. I don’t want to flame but I think I am ready to dump SIP and try IAX. It seems like a lot of work to make it work under NAT. I have posted my settings on previous posts with some helpful advice but I think I need some serious assistance. This is to be my Grad Project and I want to show that VOIP works and showcase Asterisk. I am in the Fresno, CA area. If there is anyone willing to help please PM me.
Turn on “sip set debug” on the asterisk CLI, then try making a call and pasting the debug output here.
Is this for a HS Grad Project?
This is for my College IT project.
When I call in I can see my number and CID info but nothing rings.
Here is my cli when I call:
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Mon, 18 Feb 2008 23:58:00 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
k: replaces
l: 0
Really destroying SIP dialog '4fdbe63c7a9284e13be9c6b009a95f08@sip.broadvoice.com’ Method: REGISTER
Retransmitting #3 (NAT) to 192.168.1.8:64514:
OPTIONS sip:100@192.168.1.8:64514;rinstance=c7296dee1b687727 SIP/2.0
v: SIP/2.0/UDP 192.168.1.26:5060;branch=z9hG4bK3a7d6b5c;rport
f: “asterisk” sip:asterisk@192.168.1.26;tag=as0d0db5db
t: sip:100@192.168.1.8:64514;rinstance=c7296dee1b687727
m: sip:asterisk@192.168.1.26
i: 462670bd18b739a13e170d7c79655856@192.168.1.26
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Mon, 18 Feb 2008 23:58:00 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
k: replaces
l: 0
Retransmitting #4 (NAT) to 192.168.1.8:64514:
OPTIONS sip:100@192.168.1.8:64514;rinstance=c7296dee1b687727 SIP/2.0
v: SIP/2.0/UDP 192.168.1.26:5060;branch=z9hG4bK3a7d6b5c;rport
f: “asterisk” sip:asterisk@192.168.1.26;tag=as0d0db5db
t: sip:100@192.168.1.8:64514;rinstance=c7296dee1b687727
m: sip:asterisk@192.168.1.26
i: 462670bd18b739a13e170d7c79655856@192.168.1.26
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Mon, 18 Feb 2008 23:58:00 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
k: replaces
l: 0
Really destroying SIP dialog ‘462670bd18b739a13e170d7c79655856@192.168.1.26’ Method: OPTIONS
Reliably Transmitting (NAT) to 192.168.1.8:64514:
OPTIONS sip:100@192.168.1.8:64514;rinstance=c7296dee1b687727 SIP/2.0
v: SIP/2.0/UDP 192.168.1.26:5060;branch=z9hG4bK35588351;rport
f: “asterisk” sip:asterisk@192.168.1.26;tag=as02d0147d
t: sip:100@192.168.1.8:64514;rinstance=c7296dee1b687727
m: sip:asterisk@192.168.1.26
i: 790fe2b526ef4b664c9e56e90df17654@192.168.1.26
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Mon, 18 Feb 2008 23:58:14 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
k: replaces
l: 0
Retransmitting #1 (NAT) to 192.168.1.8:64514:
OPTIONS sip:100@192.168.1.8:64514;rinstance=c7296dee1b687727 SIP/2.0
v: SIP/2.0/UDP 192.168.1.26:5060;branch=z9hG4bK35588351;rport
f: “asterisk” sip:asterisk@192.168.1.26;tag=as02d0147d
t: sip:100@192.168.1.8:64514;rinstance=c7296dee1b687727
m: sip:asterisk@192.168.1.26
i: 790fe2b526ef4b664c9e56e90df17654@192.168.1.26
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Mon, 18 Feb 2008 23:58:14 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
k: replaces
l: 0
Retransmitting #2 (NAT) to 192.168.1.8:64514:
OPTIONS sip:100@192.168.1.8:64514;rinstance=c7296dee1b687727 SIP/2.0
v: SIP/2.0/UDP 192.168.1.26:5060;branch=z9hG4bK35588351;rport
f: “asterisk” sip:asterisk@192.168.1.26;tag=as02d0147d
t: sip:100@192.168.1.8:64514;rinstance=c7296dee1b687727
m: sip:asterisk@192.168.1.26
i: 790fe2b526ef4b664c9e56e90df17654@192.168.1.26
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Mon, 18 Feb 2008 23:58:14 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
k: replaces
l: 0
Retransmitting #3 (NAT) to 192.168.1.8:64514:
OPTIONS sip:100@192.168.1.8:64514;rinstance=c7296dee1b687727 SIP/2.0
v: SIP/2.0/UDP 192.168.1.26:5060;branch=z9hG4bK35588351;rport
f: “asterisk” sip:asterisk@192.168.1.26;tag=as02d0147d
t: sip:100@192.168.1.8:64514;rinstance=c7296dee1b687727
m: sip:asterisk@192.168.1.26
i: 790fe2b526ef4b664c9e56e90df17654@192.168.1.26
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Mon, 18 Feb 2008 23:58:14 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
k: replaces
l: 0
Retransmitting #4 (NAT) to 192.168.1.8:64514:
OPTIONS sip:100@192.168.1.8:64514;rinstance=c7296dee1b687727 SIP/2.0
v: SIP/2.0/UDP 192.168.1.26:5060;branch=z9hG4bK35588351;rport
f: “asterisk” sip:asterisk@192.168.1.26;tag=as02d0147d
t: sip:100@192.168.1.8:64514;rinstance=c7296dee1b687727
m: sip:asterisk@192.168.1.26
i: 790fe2b526ef4b664c9e56e90df17654@192.168.1.26
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Mon, 18 Feb 2008 23:58:14 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
k: replaces
l: 0
Really destroying SIP dialog ‘790fe2b526ef4b664c9e56e90df17654@192.168.1.26’ Method: OPTIONS
[Feb 18 15:58:20] NOTICE[3060]: chan_sip.c:7392 sip_reregister: – Re-registration for 5598921147@sip.broadvoice.com@sip.broadvoice.com
REGISTER 13 headers, 0 lines
Reliably Transmitting (NAT) to 147.135.8.128:5060:
REGISTER sip:sip.broadvoice.com SIP/2.0
v: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK50eb69fb;rport
f: sip:5598921147@sip.broadvoice.com;tag=as43f1fdc7
t: sip:5598921147@sip.broadvoice.com
i: 4fdbe63c7a9284e13be9c6b009a95f08@sip.broadvoice.com
CSeq: 112 REGISTER
User-Agent: Asterisk PBX
Max-Forwards: 70
Authorization: Digest username=“5598921147”, realm=“BroadWorks”, algorithm=MD5, uri=“sip:sip.broadvoice.com”, nonce=“BroadWorksXfctooty0TaqvyxrBW”, response=“7330274060c66542990602310364f834”, opaque="", qop=auth, cnonce=“7670e8af”, nc=0000000a
Expires: 120
m: sip:100@127.0.0.1
o: registration
l: 0
asterisk1*CLI>
<— SIP read from 147.135.8.128:5060 —>
SIP/2.0 200 OK
Call-ID: 4fdbe63c7a9284e13be9c6b009a95f08@sip.broadvoice.com
CSeq: 112 REGISTER
From: sip:5598921147@sip.broadvoice.com;tag=as43f1fdc7
To: sip:5598921147@sip.broadvoice.com
Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK50eb69fb;received=67.187.177.30;rport=5060
Contact: sip:100@127.0.0.1
Expires: 30
Event: registration
Content-Length: 0
<------------->
— (10 headers 0 lines) —
Scheduling destruction of SIP dialog '4fdbe63c7a9284e13be9c6b009a95f08@sip.broadvoice.com’ in 6400 ms (Method: REGISTER)
[Feb 18 15:58:21] NOTICE[3060]: chan_sip.c:12475 handle_response_register: Outbound Registration: Expiry for sip.broadvoice.com is 30 sec (Scheduling reregistration in 23 s)
Really destroying SIP dialog '4fdbe63c7a9284e13be9c6b009a95f08@sip.broadvoice.com’ Method: REGISTER
Reliably Transmitting (NAT) to 192.168.1.8:64514:
OPTIONS sip:100@192.168.1.8:64514;rinstance=c7296dee1b687727 SIP/2.0
v: SIP/2.0/UDP 192.168.1.26:5060;branch=z9hG4bK793dafcc;rport
f: “asterisk” sip:asterisk@192.168.1.26;tag=as4ac0259f
t: sip:100@192.168.1.8:64514;rinstance=c7296dee1b687727
m: sip:asterisk@192.168.1.26
i: 25b3ee26787f22927ffadec2511abb26@192.168.1.26
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Mon, 18 Feb 2008 23:58:28 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
k: replaces
l: 0
Retransmitting #1 (NAT) to 192.168.1.8:64514:
OPTIONS sip:100@192.168.1.8:64514;rinstance=c7296dee1b687727 SIP/2.0
v: SIP/2.0/UDP 192.168.1.26:5060;branch=z9hG4bK793dafcc;rport
f: “asterisk” sip:asterisk@192.168.1.26;tag=as4ac0259f
t: sip:100@192.168.1.8:64514;rinstance=c7296dee1b687727
m: sip:asterisk@192.168.1.26
i: 25b3ee26787f22927ffadec2511abb26@192.168.1.26
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Mon, 18 Feb 2008 23:58:28 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
k: replaces
l: 0
Retransmitting #2 (NAT) to 192.168.1.8:64514:
OPTIONS sip:100@192.168.1.8:64514;rinstance=c7296dee1b687727 SIP/2.0
v: SIP/2.0/UDP 192.168.1.26:5060;branch=z9hG4bK793dafcc;rport
f: “asterisk” sip:asterisk@192.168.1.26;tag=as4ac0259f
t: sip:100@192.168.1.8:64514;rinstance=c7296dee1b687727
m: sip:asterisk@192.168.1.26
i: 25b3ee26787f22927ffadec2511abb26@192.168.1.26
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Mon, 18 Feb 2008 23:58:28 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
k: replaces
l: 0
Retransmitting #3 (NAT) to 192.168.1.8:64514:
OPTIONS sip:100@192.168.1.8:64514;rinstance=c7296dee1b687727 SIP/2.0
v: SIP/2.0/UDP 192.168.1.26:5060;branch=z9hG4bK793dafcc;rport
f: “asterisk” sip:asterisk@192.168.1.26;tag=as4ac0259f
t: sip:100@192.168.1.8:64514;rinstance=c7296dee1b687727
m: sip:asterisk@192.168.1.26
i: 25b3ee26787f22927ffadec2511abb26@192.168.1.26
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Mon, 18 Feb 2008 23:58:28 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
k: replaces
l: 0
Retransmitting #4 (NAT) to 192.168.1.8:64514:
OPTIONS sip:100@192.168.1.8:64514;rinstance=c7296dee1b687727 SIP/2.0
v: SIP/2.0/UDP 192.168.1.26:5060;branch=z9hG4bK793dafcc;rport
f: “asterisk” sip:asterisk@192.168.1.26;tag=as4ac0259f
t: sip:100@192.168.1.8:64514;rinstance=c7296dee1b687727
m: sip:asterisk@192.168.1.26
i: 25b3ee26787f22927ffadec2511abb26@192.168.1.26
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Mon, 18 Feb 2008 23:58:28 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
k: replaces
l: 0
Really destroying SIP dialog ‘25b3ee26787f22927ffadec2511abb26@192.168.1.26’ Method: OPTIONS
I looked at your last post. What version of Asterisk are you running? Few things you may want to change/add:
in sip.conf:
in the general section add
localnet=192.168.1.0/255.255.255.0
externip=xx.xx.xx.xx ; replace with your actual nat’d address
For your internal entries (peers 100 and 101) in sip.conf add a nat=no line and change them from type=peer to type=friend
I don’t see where you are limiting which codecs are allowed, but you didn’t make it to that part yet.
If using 1.4 change insecure=very to insecure=port, invite
The sip debug output you pasted is missing the invite messages.
Let me know if this helps.
Ok, I changed everything except the local net. I did 192.168.1.0/255.255.255.224 only because that is how I have my network setup.
Here is the invite:
[Feb 18 17:15:32] NOTICE[3060]: chan_sip.c:13859 handle_request_invite: Call from ‘5598921147’ to extension ‘5598921147’ rejected because extension not found.
Scheduling destruction of SIP dialog ‘2310281-31@147.135.8.128’ in 6400 ms (Method: INVITE)
asterisk1*CLI>
<— SIP read from 147.135.8.128:5060 —>
ACK sip:5598921147@67.187.177.30:5060 SIP/2.0
Call-ID: 2310281-31@147.135.8.128
CSeq: 1 ACK
From: "Fresno CA"sip:5599776454@147.135.8.128;user=phone;tag=xz12
To: "Eddie Granados"sip:100@67.187.177.30;tag=as6839f286
Via: SIP/2.0/UDP 147.135.8.128:5060
Content-Length: 0
<------------->
— (7 headers 0 lines) —
Really destroying SIP dialog ‘2310281-31@147.135.8.128’ Method: ACK
Really destroying SIP dialog '0c97ee067ebc7c295b68809b245b0d24@sip.broadvoice.com’ Method: REGISTER
[Feb 18 17:15:50] NOTICE[3060]: chan_sip.c:7392 sip_reregister: – Re-registration for 5598921147@sip.broadvoice.com@sip.broadvoice.com
REGISTER 13 headers, 0 lines
Reliably Transmitting (NAT) to 147.135.8.128:5060:
REGISTER sip:sip.broadvoice.com SIP/2.0
v: SIP/2.0/UDP 67.187.177.30:5060;branch=z9hG4bK72f82845;rport
f: sip:5598921147@sip.broadvoice.com;tag=as6fde84c2
t: sip:5598921147@sip.broadvoice.com
i: 0c97ee067ebc7c295b68809b245b0d24@sip.broadvoice.com
CSeq: 111 REGISTER
User-Agent: Asterisk PBX
Max-Forwards: 70
Authorization: Digest username=“5598921147”, realm=“BroadWorks”, algorithm=MD5, uri=“sip:sip.broadvoice.com”, nonce=“BroadWorksXfctrjkiwTeyc7jzBW”, response=“c2cf73f52e54752f42c08e1269978cfb”, opaque="", qop=auth, cnonce=“0a8cd4cb”, nc=00000004
Expires: 120
m: sip:100@67.187.177.30
o: registration
l: 0
asterisk1*CLI>
<— SIP read from 147.135.8.128:5060 —>
SIP/2.0 200 OK
Call-ID: 0c97ee067ebc7c295b68809b245b0d24@sip.broadvoice.com
CSeq: 111 REGISTER
From: sip:5598921147@sip.broadvoice.com;tag=as6fde84c2
To: sip:5598921147@sip.broadvoice.com
Via: SIP/2.0/UDP 67.187.177.30:5060;branch=z9hG4bK72f82845
Contact: sip:100@67.187.177.30
Expires: 30
Event: registration
Content-Length: 0
<------------->
— (10 headers 0 lines) —
Scheduling destruction of SIP dialog '0c97ee067ebc7c295b68809b245b0d24@sip.broadvoice.com’ in 6400 ms (Method: REGISTER)
[Feb 18 17:15:50] NOTICE[3060]: chan_sip.c:12475 handle_response_register: Outbound Registration: Expiry for sip.broadvoice.com is 30 sec (Scheduling reregistration in 23 s)
Really destroying SIP dialog '0c97ee067ebc7c295b68809b245b0d24@sip.broadvoice.com’ Method: REGISTER
Ok try this…
Change your [sip.broadvoice.com] peer name text to be [broadvoice], and change the type=peer to type=friend
And then change the register line to be:
register = 5598921147@sip.broadvoice.com:xxxxxxxxxxx:5598921147@sip.broadvoice.com/broadvoice
If this still does not work. Turn on normal debugging as well. So I can see further output.
core set verbose 10
core set debug 10
I tried it and I get no ringing.
cli
<— Reliably Transmitting (NAT) to 147.135.8.128:5060 —>
SIP/2.0 404 Not Found
v: SIP/2.0/UDP 147.135.8.128:5060;received=147.135.8.128
f: "Fresno CA"sip:5599776454@147.135.8.128;user=phone;tag=vxy0
t: "Eddie Granados"sip:broadvoice@67.187.177.30;tag=as6d0a542e
i: 22b0159-2b@147.135.8.128
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
k: replaces
l: 0
<------------>
[Feb 18 18:09:22] NOTICE[3060]: chan_sip.c:13859 handle_request_invite: Call from ‘5598921147’ to extension ‘5598921147’ rejected because extension not found.
Scheduling destruction of SIP dialog ‘22b0159-2b@147.135.8.128’ in 6400 ms (Method: INVITE)
asterisk1*CLI>
<— SIP read from 147.135.8.128:5060 —>
ACK sip:5598921147@67.187.177.30:5060 SIP/2.0
Call-ID: 22b0159-2b@147.135.8.128
CSeq: 1 ACK
From: "Fresno CA"sip:5599776454@147.135.8.128;user=phone;tag=vxy0
To: "Eddie Granados"sip:broadvoice@67.187.177.30;tag=as6d0a542e
Via: SIP/2.0/UDP 147.135.8.128:5060
Content-Length: 0
<------------->
— (7 headers 0 lines) —
Really destroying SIP dialog ‘22b0159-2b@147.135.8.128’ Method: ACK
asterisk1*CLI>
<— SIP read from 147.135.8.128:5060 —>
NOTIFY sip:5598921147@sip.broadvoice.com:5060 SIP/2.0
Call-ID: BW210847409180208-1673505820@192.168.0.3
CSeq: 398042264 NOTIFY
From: sip:sip.broadvoice.com;tag=35340019-1203386927409-
To: sip:5598921147@sip.broadvoice.com
Via: SIP/2.0/UDP 147.135.8.128:5060;branch=z9hG4bK-BroadWorks.192.168.0.3-192.168.8.128V5060-0-398042264-35340019-1203386927409-
Contact: sip:147.135.8.128:5060
Event: message-summary
Subscription-State:terminated
Content-Length: 43
Content-Type: application/simple-message-summary
Messages-Waiting: yes
voice-message: 5/0
<------------->
— (11 headers 2 lines) —
<— Transmitting (NAT) to 147.135.8.128:5060 —>
SIP/2.0 489 Bad event
v: SIP/2.0/UDP 147.135.8.128:5060;branch=z9hG4bK-BroadWorks.192.168.0.3-192.168.8.128V5060-0-398042264-35340019-1203386927409-;received=147.135.8.128
f: sip:sip.broadvoice.com;tag=35340019-1203386927409-
t: sip:5598921147@sip.broadvoice.com;tag=as10cb8bf7
i: BW210847409180208-1673505820@192.168.0.3
CSeq: 398042264 NOTIFY
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
k: replaces
l: 0
<------------>
asterisk1*CLI>
<— SIP read from 147.135.8.128:5060 —>
NOTIFY sip:5598921147@sip.broadvoice.com:5060 SIP/2.0
Call-ID: BW210847409180208-1673505820@192.168.0.3
CSeq: 398042264 NOTIFY
From: sip:sip.broadvoice.com;tag=35340019-1203386927409-
To: sip:5598921147@sip.broadvoice.com
Via: SIP/2.0/UDP 147.135.8.128:5060;branch=z9hG4bK-BroadWorks.192.168.0.3-192.168.8.128V5060-0-398042264-35340019-1203386927409-
Contact: sip:147.135.8.128:5060
Event: message-summary
Subscription-State:terminated
Content-Length: 43
Content-Type: application/simple-message-summary
Messages-Waiting: yes
voice-message: 5/0
<------------->
— (11 headers 2 lines) —
<— Transmitting (NAT) to 147.135.8.128:5060 —>
SIP/2.0 489 Bad event
v: SIP/2.0/UDP 147.135.8.128:5060;branch=z9hG4bK-BroadWorks.192.168.0.3-192.168.8.128V5060-0-398042264-35340019-1203386927409-;received=147.135.8.128
f: sip:sip.broadvoice.com;tag=35340019-1203386927409-
t: sip:5598921147@sip.broadvoice.com;tag=as794635b6
i: BW210847409180208-1673505820@192.168.0.3
CSeq: 398042264 NOTIFY
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
k: replaces
l: 0
<------------>
[Feb 18 18:09:35] NOTICE[3060]: chan_sip.c:7392 sip_reregister: – Re-registration for 5598921147@sip.broadvoice.com@sip.broadvoice.com
REGISTER 12 headers, 0 lines
REGISTER attempt 1 to 5598921147@sip.broadvoice.com@sip.broadvoice.com
Reliably Transmitting (NAT) to 147.135.8.128:5060:
REGISTER sip:sip.broadvoice.com SIP/2.0
v: SIP/2.0/UDP 67.187.177.30:5060;branch=z9hG4bK5c33e97a;rport
f: sip:5598921147@sip.broadvoice.com;tag=as32d11753
t: sip:5598921147@sip.broadvoice.com
i: 1ab232662b65822f580d033539681c2d@127.0.0.1
CSeq: 139 REGISTER
User-Agent: Asterisk PBX
Max-Forwards: 70
Expires: 120
m: sip:broadvoice@67.187.177.30
o: registration
l: 0
asterisk1*CLI>
<— SIP read from 147.135.8.128:5060 —>
SIP/2.0 200 OK
Call-ID: 1ab232662b65822f580d033539681c2d@127.0.0.1
CSeq: 139 REGISTER
From: sip:5598921147@sip.broadvoice.com;tag=as32d11753
To: sip:5598921147@sip.broadvoice.com
Via: SIP/2.0/UDP 67.187.177.30:5060;branch=z9hG4bK5c33e97a
Contact: sip:broadvoice@67.187.177.30
Expires: 30
Event: registration
Content-Length: 0
<------------->
— (10 headers 0 lines) —
Scheduling destruction of SIP dialog ‘1ab232662b65822f580d033539681c2d@127.0.0.1’ in 32000 ms (Method: REGISTER)
[Feb 18 18:09:35] NOTICE[3060]: chan_sip.c:12475 handle_response_register: Outbound Registration: Expiry for sip.broadvoice.com is 30 sec (Scheduling reregistration in 23 s)
Still no ringing. I will add my extensions.conf also
extensions.conf
[outgoing]
exten => _1NXXNXXXXXX, 1, dial(SIP/${EXTEN}@sip.broadvoice.com,30)
exten => _1NXXNXXXXXX, 2, congestion()
exten => _1NXXNXXXXXX, 102, busy()
; This extended Dial Plan will enable International Dialing on The Unlimited World PLUS Plan
; This dial plan enables World Plus countries
; there are no built in ways to prevent calls to cell phone users (except in germany where Cell phone prefix’s are
; carried by 1 and has been accounted for)
exten = _01130.,1,dial(SIP/${EXTEN}@sip.broadvoice.com,30)
exten = _01131.,1,dial(SIP/${EXTEN}@sip.broadvoice.com,30)
exten = _01132.,1,dial(SIP/${EXTEN}@sip.broadvoice.com,30)
exten = _01133.,1,dial(SIP/${EXTEN}@sip.broadvoice.com,30)
exten = _01134.,1,dial(SIP/${EXTEN}@sip.broadvoice.com,30)
exten = _011351.,1,dial(SIP/${EXTEN}@sip.broadvoice.com,30)
exten = _011352.,1,dial(SIP/${EXTEN}@sip.broadvoice.com,30)
exten = _011353.,1,dial(SIP/${EXTEN}@sip.broadvoice.com,30)
exten = _011378.,1,dial(SIP/${EXTEN}@sip.broadvoice.com,30)
exten = _01139.,1,dial(SIP/${EXTEN}@sip.broadvoice.com,30)
exten = _01141.,1,dial(SIP/${EXTEN}@sip.broadvoice.com,30)
exten = _011420.,1,dial(SIP/${EXTEN}@sip.broadvoice.com,30)
exten = _01143.,1,dial(SIP/${EXTEN}@sip.broadvoice.com,30)
exten = _01144.,1,dial(SIP/${EXTEN}@sip.broadvoice.com,30)
exten = _01145.,1,dial(SIP/${EXTEN}@sip.broadvoice.com,30)
exten = _01146.,1,dial(SIP/${EXTEN}@sip.broadvoice.com,30)
exten = _01147.,1,dial(SIP/${EXTEN}@sip.broadvoice.com,30)
exten = _01148.,1,dial(SIP/${EXTEN}@sip.broadvoice.com,30)
exten = _01149[2-9].,1,dial(SIP/${EXTEN}@sip.broadvoice.com,30)
exten = _01154.,1,dial(SIP/${EXTEN}@sip.broadvoice.com,30)
exten = _01155.,1,dial(SIP/${EXTEN}@sip.broadvoice.com,30)
exten = _01156.,1,dial(SIP/${EXTEN}@sip.broadvoice.com,30)
exten = _01160.,1,dial(SIP/${EXTEN}@sip.broadvoice.com,30)
exten = _01161.,1,dial(SIP/${EXTEN}@sip.broadvoice.com,30)
exten = _01164.,1,dial(SIP/${EXTEN}@sip.broadvoice.com,30)
exten = _01165.,1,dial(SIP/${EXTEN}@sip.broadvoice.com,30)
exten = _01181.,1,dial(SIP/${EXTEN}@sip.broadvoice.com,30)
exten = _01182.,1,dial(SIP/${EXTEN}@sip.broadvoice.com,30)
exten = _011852.,1,dial(SIP/${EXTEN}@sip.broadvoice.com,30)
exten = _01186.,1,dial(SIP/${EXTEN}@sip.broadvoice.com,30)
exten = _011886.,1,dial(SIP/${EXTEN}@sip.broadvoice.com,30)
exten = _011972.,1,dial(SIP/${EXTEN}@sip.broadvoice.com,30)
exten = _011.,2,congestion() ; No answer, nothing
exten = _011.,102,busy() ; Busy
[asterisk_guitools]
exten = executecommand,1,System(${command})
exten = executecommand,n,Hangup()
exten = record_vmenu,1,Answer
exten = record_vmenu,n,Playback(vm-intro)
exten = record_vmenu,n,Record(${var1})
exten = record_vmenu,n,Playback(vm-saved)
exten = record_vmenu,n,Playback(vm-goodbye)
exten = record_vmenu,n,Hangup
exten = play_file,1,Answer
exten = play_file,n,Playback(${var1})
exten = play_file,n,Hangup
hasbeensetup = Y
[timebasedrules]
[phones] ;This is the context from the Sip.conf i
nclude => internal
include => outgoing
[numberplan-custom-1]
plancomment = DialPlan1
include = default
include = parkedcalls
[globals]
CONSOLE = Console/dsp
IAXINFO = guest
TRUNK = Zap/g2
TRUNKMSD = 1
include = daytime|9:00-17:00|mon-fri||
include = weekend||sat-sun||*
include = weeknights|17:02-8:58|mon-fri||
Eddie = SIP/100
Elaina = SIP/101
Val = SIP/102
[from-broadvoice]
exten = 100,1,Answer()
exten = 100,n,Background(enter-ext-of-person)
exten = 100,n,WaitExten()
exten = 1,1,Dial (${Eddie},30)
exten = 1,n,Playback(vm-nobodyavail)
exten = 1,n,Hangup()
[internal]
exten = 100,1,Dial (${Eddie})
exten = eddie,1,Dial(${Eddie})
exten = 101,1,Dial (${Elaina})
exten = elaina,1,Dial (${Elaina})
exten = 102,1,Dial (${Val})
exten = val,1,Dial (${Val})
exten = 850,1,VoiceMailMain()
Ok that is a little different than the one I saw I the other thread.
In the context set by the broadvoice peer, add the extension 5598921147. The other thread I was looking at from you had it set.
I got it. I did what you asked and then I looked at another Broadvoice forum and took the best of both worlds. I am able to call in and out. Thank you so much. I will post the context this evening for other people dealing with Broadvoice. I will add you to my References section in my grad project book. Please PM me your info so I can give credit where credit is due.